Calls not working inbound

Asterisk Asterisk Support
1

I’m having a problem, Calls not coming in but I do have some events

Reliably Transmitting (no NAT) to 10.117.220.15:5060:
OPTIONS sip:10.117.220.15 SIP/2.0
Via: SIP/2.0/UDP 10.117.220.11:5060;branch=z9hG4bK2555b958
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.117.220.11;tag=as3fb79988
To: sip:10.117.220.15
Contact: sip:asterisk@10.117.220.11:5060
Call-ID: 249e2fce61d768f96b781bf810234fba@10.117.220.11:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.25.1
Date: Wed, 10 Feb 2021 22:52:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.117.220.15:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.117.220.11:5060;branch=z9hG4bK2555b958
From: “asterisk” sip:asterisk@10.117.220.11;tag=as3fb79988
To: sip:10.117.220.15;tag=34B140-1818
Date: Thu, 11 Feb 2021 09:52:44 GMT
Call-ID: 249e2fce61d768f96b781bf810234fba@10.117.220.11:5060
Server: Cisco-SIPGateway/IOS-15.4.3.M9
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 326

v=0
o=CiscoSystemsSIP-GW-UserAgent 827 5325 IN IP4 10.117.220.15
s=SIP Call
c=IN IP4 10.117.220.15
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
m=image 0 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
— (14 headers 13 lines) —
Really destroying SIP dialog ‘249e2fce61d768f96b781bf810234fba@10.117.220.11:5060’ Method: OPTIONS

<— SIP read from UDP:10.117.220.15:5060 —>
OPTIONS sip:X:5060 SIP/2.0
Via: SIP/2.0/UDP 10.117.220.15:5060;branch=z9hG4bK3A1D1F1E77
From: sip:10.117.220.15;tag=34F3DC-E53
To: sip:X
Date: Thu, 11 Feb 2021 09:53:01 GMT
Call-ID: 8F8EBC9B-6B2911EB-9EDBEF98-F7E60E88@10.117.220.15
User-Agent: Cisco-SIPGateway/IOS-15.4.3.M9
Max-Forwards: 70
CSeq: 101 OPTIONS
Contact: sip:10.117.220.15:5060
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 10.117.220.15:5060 (no NAT)
Looking for s in gateways-inbound (domain X)

<— Transmitting (no NAT) to 10.117.220.15:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.117.220.15:5060;branch=z9hG4bK3A1D1F1E77;received=10.117.220.15
From: sip:10.117.220.15;tag=34F3DC-E53
To: sip:X;tag=as3dba45fd
Call-ID: 8F8EBC9B-6B2911EB-9EDBEF98-F7E60E88@10.117.220.15
CSeq: 101 OPTIONS
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:10.117.220.11:5060
Accept: application/sdp
Content-Length: 0

2

There are no incoming (or outgoing) call attempts (INVITE) in that log. On the evidence provided, there are no incoming calls because none ever reach Asterisk.

Also, please markup logs (and configurations as pre-formatted text, otherwise they get garbled by the forum).

3

I had a dns issue going on

4

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